Communication networks consist of interconnected nodes and can be subdivided into core networks and access networks, the latter providing access to user equipment, for example a wireless access for mobile user equipment to a radio access network. Core networks interconnect access networks and further networks, e.g. other core networks or the Internet. In the Universal Mobile Telecommunications System (UMTS) architecture, an access network can be controlled by a Radio Network Controller (RNC), which is connected to the core network and provides access to the core network, i.e. serves as access node. In the Global System for Mobile Communications (GSM) architecture, the access network is controlled by a Base Station Controller (BSC). The 3G core network is controlled by one or more Mobile Switching Centres (MSCs). These MSCs also influence the decisions in RNC and BSC.
For the transmission on a connection, speech (or other media) is encoded (and subsequently decoded) according to one or more encoding/decoding schemes, also referred to herein as coding schemes and alternatively denoted “codecs”. Determination of an optimal codec or set of codecs may be done by means of Codec Negotiation. A coding scheme can transport speech either in a compressed or in a non-compressed mode. In many networks, different coding schemes can be used and different nodes can have different capabilities for handling the coding schemes. Speech transcoders perform the transcoding between different speech coding schemes, i.e. they decode the one scheme into speech (linear PCM or other representation) and then encode the speech by the other scheme. Hence, a transcoder is a device that performs a codec, i.e. it implements a particular coding scheme (in fact a transcoder can implement a number of coding schemes and employ them on per call basis as requested by call/session control applications). Tandem Free Operation (TFO) is a configuration of two transcoders with compatible coding schemes on the compressed voice sides at both ends of a connection, i.e. on the interface to the user equipment. In this case, the transcoding stages can be bypassed and the compressed voice coding is used end to end in the connection (see 3GPP TS 28.062).
Out of Band Transcoder Control (OoBTC) permits speech connections to be established end to end with a common coding scheme, i.e. ideally the same speech coding is used in the whole connection between the access networks. The advantage is that maintaining compressed voice saves core network bandwidth and optimizes speech quality, because transcoding stages, which in principle always introduce distortion, are avoided (see 3GPP TS 23.153).
An International Telecommunication Union (ITU) protocol called Bearer Independent Call Control (BICC) supports out of band signaling procedures, which allow a negotiation of the coding scheme between network nodes. In the ITU-Telecommunication Standardisation Sector (ITU-T) proposal BICC Q. 1901 (ITU, June 2000), coding scheme negotiation is performed from the originating control node in a connection to each subsequent node by including a list of allowed coding schemes in the Application Transport Parameter (APP) parameter in the Initial Address Message (IAM) for the set-up of the connection. Each node checks the list and if it does not support a particular coding type it removes it from the list. The adapted list is passed on with the JAM and any non-supported types are removed as long as the BICC signaling is supported. When the final node, either the terminating node or the last node supporting BICC, is reached, the coding scheme type is selected by the node. This selected coding scheme and the list of remaining, commonly supported codec schemes are returned to the originating node via all intermediate nodes.
In the BICC coding scheme negotiation procedures there are no rules for defining how many transcoder stages are allowed and whether an access network that supports out of band coding scheme negotiation can activate transcoders to keep Transcoder Free Operation (TrFO) between the access node and the rest of the network. The number of transcoding stages in a connection end to end can significantly affect the speech quality. More than three transcoding stages typically cause substantial speech impairment. The number of stages causing a substantial impairment depends on the coding algorithm/scheme and the speech impairment by further entities in the connection.
The coding scheme negotiation procedures may result in transcoders being activated to enable supplementary services or because the bearer technology in a node or network does not support compressed voice. For example, Asynchronous Transfer Mode (ATM) networks allow transmission of either compressed or non-compressed speech, while Synchronous Transfer Mode (STM) networks require non-compressed speech coding, which via bit stealing can include TFO with compressed speech (TFO is not really required in STM). Furthermore, the negotiation should result in the optimum location of the transcoders, which is with today's technologies not always the case. For example, for connections exiting a STM network to ATM, a transcoder should be located at the network edge to save bandwidth in the ATM network by use of a compressed coding scheme.
In many cases, it is necessary to modify the coding scheme in a section of a connection. For example, a connection is often transferred between different access networks due to a handover. Modifications in the core network are disadvantageous, especially if they require increased transmission bandwidth, which will sometimes not be available causing a termination of the connection. The number of transcoder stages in a connection can be increased by a modification, with corresponding quality impairment. Again, an optimum location of transcoders is often not achieved.
These issues will now be described in greater detail with reference to the examples of FIGS. 1-4. FIG. 1 illustrates a speech call originating in Integrated Services Digital Network (ISDN), with pulse code modulation (PCM) A-law coding immediately inside an ISDN-terminal. (A-law is an ITU standard (G.711) for converting analog data into digital form using PCM.) For simplicity, the local exchange that controls the ISDN access is not shown. The TSC shown therein is a transit switching server, which may be collocated with an MSC. The call is routed to a Bearer Independent Core Network (BICN), where Codec Negotiation using Out-of-Band Transcoder Control (OoBTC) is supported, and the call terminates in a mobile terminal on a GSM access. For this example, it is assumed that the BICN and the GSM access support all state-of-the-art GSM Codec Types in TFO. The mobile terminal also supports all GSM Codec Types. The optimal Codec Selection is then to use the best Codec Type supported by the mobile terminal, e.g. the Full Rate—Adaptive Multirate (FR_AMR) codec. The unavoidable transcoder (PCM A-law to FR_AMR) is placed at the interconvection from ISDN to the BICN, i.e. close to the originating side. Optimal voice quality (under these conditions) is achieved with minimal bit rate in the BICN. In this example, the Codec Negotiation assumes that the ISDN access has only used PCM A-law and no other transcoding before. The offered “Supported Codec List” in OoBTC in this case offers PCM A-law as first priority, without any additional information associated with this Codec entry. Other Codec entries follow after this, for example the FR_AMR. The terminating side of the Codec negotiation knows that compression is necessary inside the GSM mobile terminal, but that it is unimportant from a voice quality point of view where to place the transcoder.
FIG. 2 illustrates the same speech call, from ISDN to a GSM mobile, but in this case the mobile is not state-of-the-art and the only Codec Type supported is GSM—Full Rate (GSM_FR) for which, in this case, the GSM radio access does not support TFO. Then “PCM A-law” inside the BICN is optimal for optimal voice quality, although the bit rate is substantially higher. Also in the example of FIG. 2, the Codec Negotiation assumes that the ISDN access has only used PCM A-law but no previous transcoding. Again the offered “Supported Codec List” in OoBTC in this case offers PCM A-law as first priority, without any additional information associated with this codec entry. Other codec entries follow after this, for example the FR_AMR. The terminating node has all necessary information on the terminating radio access and can perform the optimal decision. For example, if voice quality is most important, the terminating radio access determines that FR_AMR is not a good choice.
FIG. 3 illustrates the same scenario, but where bit rate through the BICN is to be minimized (instead of maximizing voice quality). Hence, in the example of FIG. 3, the terminating radio access instead selects FR_AMR for the BICN link and GSM_FR for the GSM access.
FIG. 4 illustrates a scenario wherein AMR is supported at the terminating end, but without TFO support. The terminating side can determine, however, that AMR has a very high quality, much higher than conventional GSM_FR. In this example, the operator puts a high weight on bit rate saving in the core network. Hence, AMR is also selected for the Core network in this case, although, in this case, an additional transcoding stage (i.e. a pair of transcoders) must be activated. In particular, compare FIG. 4 with FIG. 1 discussed above.
Now, considering the four examples together, the example of FIG. 1 achieves the best quality, where only one transcoding from AMR to PCM is performed, because AMR is the optimal codec in this case. The example of FIG. 3 achieves the worst quality, because GSM_FR is substantially worse than AMR and the additional transcoding to AMR for bit rate saving degrades an already compromised voice quality too much (at least for some operators). The example of FIG. 2 has exactly the number of transcoding stages as in the example of FIG. 1, but the voice quality is much lower than in the example of FIG. 1, but better than in the example of FIG. 3. In the example of FIG. 4, a second transcoding stage has been added as compared with the example of FIG. 3. The resulting subjective speech quality is better than in the example of FIG. 3 and even better than in the example of FIG. 2 due to the fact that the use of two AMRs in sequence is still better than the use of GSM_FR once.
These conclusions are derived using otherwise conventional E-Model analysis techniques, which assign an “impairment” factor of “+1” to PCM A-law, “+20” to GSM_FR, and “+5” to AMR (12.2). E-model is a computational model for use in transmission planning. AMR (12.2) represents one particular AMR codec mode. Conventional E-Model techniques assume that these impairments are added along the voice path. For the particular examples of FIGS. 1-4, the following impairment can then be calculated:                Example of FIG. 1: Impairment=1+5=6 transcoding stages: 1        Example of FIG. 2: Impairment=1+20=21 transcoding stages: 1        Example of FIG. 3: Impairment=1+5+1+20=27 transcoding stages: 3        Example of FIG. 4: Impairment=1+5+1+5=12 transcoding stages: 3        
Note that, for some communication system operators, it might be acceptable to allow additional speech compression for bit rate saving, but only in the example of FIG. 4, but not in example of FIG. 3. This can be verified by using otherwise conventional Perceptual Evaluation of Speech Quality (PESQ) tools. PESQ sends a carefully selected speech utterance through an established voice path (or a simulated model of it) and records the resulting output signal. PESQ then compares the input utterance with the output signal and calculates the impairment.
In view of the foregoing, it would be highly desirable to provide techniques for allowing network nodes to distinguish between the various scenarios of FIGS. 1-4 to identify, on a call-by-call basis, the optimal selection of codecs to be activated into a connection. However, conventional Codec Negotiation protocols (i.e. TFO, OoBTC, Session Initiation Protocol (SIP) and Session Description Protocol (SDP), Internet Protocol Multimedia Subsystem (IMS), or ISDN User Part (ISUP)), do not provide information on how many transcoding stages are already within the speech path and not how much the speech quality is already degraded. This information is, however, required for the optimal selection, as shown in the examples above.
These and other problems were initially addressed by PCT Patent Application WO 02/32152, entitled “Method and Node for the Control of a Connection in a Communication Network”, of Ericsson Telefon AB L M. Briefly, that patent application describes a technique wherein an indicator is forward among nodes of a communication network that identifies, e.g., the number of speech transcoders present along a connection or the accumulated speech impairment along the connection. Nodes controlling the connection use the indicator to determine whether to activate or deactivate speech transcoders along the connection. For example, if the indicator indicates that no transcoders are present in the connection, a transcoder can be advantageously added. On the other hand, if the indicator indicates that one or more transcoders are already present, then preferably no additional transcoders should be added.
In one example described in WO 02/32152, the indicator is a merely flag indicating whether at least one speech transcoder is present in the connection. This allows a simple implementation of the technique utilizing small message size. In another example, the indicator is a counter indicating the number of speech transcoders in the connection. In still other examples, the indicator is a variable indicating the accumulated speech impairment by speech transcoders in the connection, which is compared again one or more numerical thresholds to evaluate optimal transcoder arrangements.
By exploiting the information contained within the indicator, the technique of WO 02/32152 allows for improved selection of transcoders on a call-by-call basis to achieve enhancement of the average quality of connections in a communication network while avoiding deterioration of a connection due to changes in the coding scheme. In other words, nodes can intelligently exploit the information contained within the indicator to make informed decisions regarding modifications to network connections, particularly the activation or deactivation of codecs or other transcoders. Moreover, any impact on the connection in a core network is minimized because many modifications can be kept local in a single node or in an adjacent pair of nodes. Additionally, the techniques are not limited to speech transcoders but are more generally applicable to any entities affecting connection quality. Other examples of such entities include conference devices for connecting conference calls.
Although the technique of WO 02/32152 represents a significant improvement of previous techniques, room for further improvement remains. For example, whereas the indicator of WO 02/32152 can provide an indication of the accumulated (speech) impairment along a connection, it would be beneficial to provide information pertaining to the speech impairment or other impairment arising due to each individual coding scheme of a telecommunication service payload, transcoder or other entity affecting connection quality, so as to permit a more informed decision.
A further complication arises due to the fact that a given speech coding scheme (compression algorithm such as AMR) can be applied on different links with different link characteristics. An AMR compression on a GSM radio link with transmission errors has substantially different overall quality impairments compared to an AMR on a fixed link without errors. Even between radio links the impairment is different: an AMR (7.4 kbit/s) of a GSM Full Rate Channel is substantially more error robust than on a GSM Half Rate Channel. AMR on an UMTS radio channel has yet another impairment. The 3GPP standard TS 26.103 addresses these different radio access characteristics by differentiating the “Codec Type” entry in the Codec List by compression algorithm (“AMR”, “EFR”) and by radio access “FR_”, “HR_”, “OHR_”, “GSM_” or “UMTS_”). However the Session Description Protocol/Session Initiation Protocol (SDP/SIP) does not take this differentiation into account (e.g. the SDP/SIP maps all AMR Codec Types into a generic “AMR”) and is in that respect missing information to provide the basis for a good decision. Even worse, radio impairments also depend heavily on the radio network design and the actual radio conditions. Some of these radio impairments are dynamically varying over time and location, others are semi-static. Moreover, further impairments may occur along the speech path, e.g. conference devices, echo suppressors, noise reduction devices and many more, which need to be known for the optimal Codec Selection.
Accordingly, it would be desirable to provide still further information within a connection impairment indicator so as to allow nodes to make more informed decisions. It is to this end that the present invention is directed.